Always try to use the latest WebRTC API with the latest Asterisk branch(11 or 12). Be sure you have the icessuport enabled in the rtp.conf

ISSUE: I get this response on JSSIP or SIPML5 debug:tRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd.This issue is caused because you asterisk don't have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your system. Then recompile asterisk(be sure to rerun the configure script before the make command).

ISSUE: I get this response on JSSIP or SIPML5:Failed to get local SDPThis issue can be caused as the same cuase described above, be sure you aren't getting the above message in the javascript debug.

If you aren't getting the lack of ice-ufrag and ice-pwd then check your peer settings in the asterisk side. Remember you need in your peer settings:-- avpf=yes-- icesupport=yes-- encryption=yes-- transport=ws

And disable the videosupport if you don't set the use of the vide in the WebRTC API.

ISSUE: I get this in the Asterisk cli Rejecting secure audio stream without encryption details: audio RTP/SAVPF: Be sure that each webrtc peer have in the settings these values:-- avpf=yes-- encryption=yes

Setting that globally doesn't work.

ISSUE:Not getting audio at allUsually this is caused by two reasons: ** Wrong NAT settings in your Asterisk side. ** The ICE server is setting the wrong IP to use in the SDP.

Fixing the NAT settings:First check how are you making your tests in your enviroment: If you are in the same LAN then your webrtc peers no need NAT at all, so define the setting nat=no.

If you are connecting your peer to a NATed Asterisk box then define the setting nat=force_rpot,comedia.

Workaround for the ICE issue:If setting the nat settings correctly doesn't solve your issue then you need to get the Debug on Asterisk and Chrome. On the Asterisk side you need to enable the SIP debug by running the command:

ISSUE:I have all settings correct but I don't have audioCheck that your asterisk is sending the Audio to the correct IP using ICE. Enable the RTP debug in asterisk and be sure that you see an output like this:

If you are using Doubango's media gateway(WebRTC2SIP) then disable in the peer configuration the setting:avpf=yesencription=yesicesupport=yes

ISSUEI have Installed libuuid/uuid but asterisk never send the ice-ufrag & ice-pwd in the SDPThere is a bug report for Branch 11.7 to 11.8 https://issues.asterisk.org/jira/browse/ASTERISK-23425 where Asterisk never send those settings Upgrade asterisk to 11.9 and it should work out of the box.

----------------------Added 4-June-2014ISSUEAfter upgrade Chrome to the version 35 SIPML5 stop workingAs all we know Chrome dropped the support of SDES as default and now use DTLS-SRTP, you can follow the thread in the Asterisk Jira page and see the progress with the patches.

12.- Test and enjoy! If you have a supermachine this can be done in almost 40mins YAY!

Here http://pastebin.com/hJhnniBD you can take a look on a complete SIP DEBUG for a successfull call between a SIPML5 Client and a PSTN Number including the RTP DEBUG with the "VIA ICE" text and the DTLS support.

This my last attempt to help the community about WebRTC and Asterisk. Please avoid send me private messages, tweets or emails READ carefully the firts part in this thread: viewtopic.php?f=1&t=90167

Almost all scenarios are described there, Why I know that? Because I have experienced those issues too.

I don't use JsSIP because the devs are kind annoying so if you want support for JsSIP ask them and hope they don't kick you out as me and others.

Stop complaining about the month or the week you have spent on this matter, we already have like a year or more dealing with it and your complians.

If you want ASAP response or help HIRE SOMEONE. WebRTC and Asterisk aren't for N00bs, yes you have used linux and you can make web pages but you need more than that.

If you are wondering if I provide custom installation for you, the answer is YES this is my fee for a complete setup: $2000USD paypal in front.

And again STOP send me emails, privates and tweets this is annoying.

Finally be gratefull with the people involved in this, the people in the JIRA PAGE(https://issues.asterisk.org/jira/browse/ASTERISK-22961) whom make this possible and of course the Asterisk Devs like Joshua Colp, Mamadou from Doubango, etc they are the Real MVP :'( of this.

Good Luck! And READ READ and finally READ again this forum, google and the doubango discuss group.